March 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: mediatek: btcvsd add loopback
by KaiChieh Chuang 04 Mar '19

04 Mar '19

add direct loopback path from rx to tx Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 69 +++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index 1b8bcdaf..9a163d7 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -49,6 +49,7 @@ enum bt_sco_state { BT_SCO_STATE_IDLE, BT_SCO_STATE_RUNNING, BT_SCO_STATE_ENDING, + BT_SCO_STATE_LOOPBACK, }; enum bt_sco_direct { @@ -486,7 +487,8 @@ static irqreturn_t mtk_btcvsd_snd_irq_handler(int irq_id, void *dev) if (bt->rx->state != BT_SCO_STATE_RUNNING && bt->rx->state != BT_SCO_STATE_ENDING && bt->tx->state != BT_SCO_STATE_RUNNING && - bt->tx->state != BT_SCO_STATE_ENDING) { + bt->tx->state != BT_SCO_STATE_ENDING && + bt->tx->state != BT_SCO_STATE_LOOPBACK) { dev_warn(bt->dev, "%s(), in idle state: rx->state: %d, tx->state: %d\n", __func__, bt->rx->state, bt->tx->state); goto irq_handler_exit; @@ -512,6 +514,42 @@ static irqreturn_t mtk_btcvsd_snd_irq_handler(int irq_id, void *dev) buf_cnt_tx = btsco_packet_info[packet_type][2]; buf_cnt_rx = btsco_packet_info[packet_type][3]; + if (bt->tx->state == BT_SCO_STATE_LOOPBACK) { + u8 *src, *dst; + unsigned long connsys_addr_rx, ap_addr_rx; + unsigned long connsys_addr_tx, ap_addr_tx; + + connsys_addr_rx = *bt->bt_reg_pkt_r; + ap_addr_rx = (unsigned long)bt->bt_sram_bank2_base + + (connsys_addr_rx & 0xFFFF); + + connsys_addr_tx = *bt->bt_reg_pkt_w; + ap_addr_tx = (unsigned long)bt->bt_sram_bank2_base + + (connsys_addr_tx & 0xFFFF); + + if (connsys_addr_tx == 0xdeadfeed || + connsys_addr_rx == 0xdeadfeed) { + /* bt return 0xdeadfeed if read reg during bt sleep */ + dev_warn(bt->dev, "%s(), connsys_addr_tx == 0xdeadfeed\n", + __func__); + goto irq_handler_exit; + } + + src = (u8 *)ap_addr_rx; + dst = (u8 *)ap_addr_tx; + + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_BT2ARM, src, + bt->tx->temp_packet_buf, + packet_length, + packet_num); + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_ARM2BT, + bt->tx->temp_packet_buf, dst, + packet_length, + packet_num); + bt->rx->rw_cnt++; + bt->tx->rw_cnt++; + } + if (bt->rx->state == BT_SCO_STATE_RUNNING || bt->rx->state == BT_SCO_STATE_ENDING) { if (bt->rx->xrun) { @@ -1067,6 +1105,33 @@ static int btcvsd_band_set(struct snd_kcontrol *kcontrol, return 0; } +static int btcvsd_loopback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + bool lpbk_en = bt->tx->state == BT_SCO_STATE_LOOPBACK; + + ucontrol->value.integer.value[0] = lpbk_en; + return 0; +} + +static int btcvsd_loopback_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + if (ucontrol->value.integer.value[0]) { + mtk_btcvsd_snd_set_state(bt, bt->tx, BT_SCO_STATE_LOOPBACK); + mtk_btcvsd_snd_set_state(bt, bt->rx, BT_SCO_STATE_LOOPBACK); + } else { + mtk_btcvsd_snd_set_state(bt, bt->tx, BT_SCO_STATE_RUNNING); + mtk_btcvsd_snd_set_state(bt, bt->rx, BT_SCO_STATE_RUNNING); + } + return 0; +} + static int btcvsd_tx_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1202,6 +1267,8 @@ static int btcvsd_tx_timestamp_get(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new mtk_btcvsd_snd_controls[] = { SOC_ENUM_EXT("BTCVSD Band", btcvsd_enum[0], btcvsd_band_get, btcvsd_band_set), + SOC_SINGLE_BOOL_EXT("BTCVSD Loopback Switch", 0, + btcvsd_loopback_get, btcvsd_loopback_set), SOC_SINGLE_BOOL_EXT("BTCVSD Tx Mute Switch", 0, btcvsd_tx_mute_get, btcvsd_tx_mute_set), SOC_SINGLE_BOOL_EXT("BTCVSD Tx Irq Received Switch", 0, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: hdmi-codec: fix S/PDIF DAI
by Russell King 04 Mar '19

04 Mar '19

When using the S/PDIF DAI, there is no requirement to call snd_soc_dai_set_fmt() as there is no DAI format definition that defines S/PDIF. In any case, S/PDIF does not have separate clocks, this is embedded into the data stream. Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt to configure TDA998x via the hw_params callback fails as the hdmi_codec_daifmt is left initialised to zero. Since the S/PDIF DAI will only be used by S/PDIF, prepare the hdmi_codec_daifmt structure for this format. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/codecs/hdmi-codec.c | 118 +++++++++++++++++++++--------------------- 1 file changed, 59 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index e5b6769b9797..d5f73c837281 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -529,73 +529,71 @@ static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); struct hdmi_codec_daifmt cf = { 0 }; - int ret = 0; dev_dbg(dai->dev, "%s()\n", __func__); - if (dai->id == DAI_ID_SPDIF) { - cf.fmt = HDMI_SPDIF; - } else { - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - cf.bit_clk_master = 1; - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFM: - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBM_CFS: - cf.bit_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } + if (dai->id == DAI_ID_SPDIF) + return 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - cf.frame_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_NF: - cf.bit_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_IF: - cf.frame_clk_inv = 1; - cf.bit_clk_inv = 1; - break; - } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - cf.fmt = HDMI_I2S; - break; - case SND_SOC_DAIFMT_DSP_A: - cf.fmt = HDMI_DSP_A; - break; - case SND_SOC_DAIFMT_DSP_B: - cf.fmt = HDMI_DSP_B; - break; - case SND_SOC_DAIFMT_RIGHT_J: - cf.fmt = HDMI_RIGHT_J; - break; - case SND_SOC_DAIFMT_LEFT_J: - cf.fmt = HDMI_LEFT_J; - break; - case SND_SOC_DAIFMT_AC97: - cf.fmt = HDMI_AC97; - break; - default: - dev_err(dai->dev, "Invalid DAI interface format\n"); - return -EINVAL; - } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; } hcp->daifmt[dai->id] = cf; - return ret; + return 0; } static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) @@ -792,8 +790,10 @@ static int hdmi_codec_probe(struct platform_device *pdev) i++; } - if (hcd->spdif) + if (hcd->spdif) { hcp->daidrv[i] = hdmi_spdif_dai; + hcp->daifmt[DAI_ID_SPDIF].fmt = HDMI_SPDIF; + } dev_set_drvdata(dev, hcp); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: ab8500: Mark expected switch fall-through
by Gustavo A. R. Silva 04 Mar '19

04 Mar '19

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. This patch fixes the following warning: In file included from sound/soc/codecs/ab8500-codec.c:24: sound/soc/codecs/ab8500-codec.c: In function ‘ab8500_codec_set_dai_fmt’: ./include/linux/device.h:1485:2: warning: this statement may fall through [-Wimplicit-fallthrough=] _dev_err(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/ab8500-codec.c:2129:3: note: in expansion of macro ‘dev_err’ dev_err(dai->component->dev, ^~~~~~~ sound/soc/codecs/ab8500-codec.c:2132:2: note: here default: ^~~~~~~ Warning level 3 was used: -Wimplicit-fallthrough=3 This patch is part of the ongoing efforts to enable -Wimplicit-fallthrough. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/ab8500-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 03bbbcd3b6c1..87616b126018 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2129,6 +2129,7 @@ static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_err(dai->component->dev, "%s: ERROR: The device is either a master or a slave.\n", __func__); + /* fall through */ default: dev_err(dai->component->dev, "%s: ERROR: Unsupporter master mask 0x%x\n", -- 2.21.0

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[alsa-devel] [PATCH 0/5] ASoC: stm32: sai: miscellaneous fixes
by Olivier Moysan 04 Mar '19

04 Mar '19

This patch-set is a collection of fixes for STM32 SAI driver. Olivier Moysan (5): ASoC: stm32: sai: fix iec958 controls indexation ASoC: stm32: sai: fix exposed capabilities in spdif mode ASoC: stm32: sai: fix race condition in irq handler ASoC: stm32: sai: fix oversampling mode ASoC: stm32: sai: fix set_sync service sound/soc/stm/stm32_sai.c | 7 ++++--- sound/soc/stm/stm32_sai_sub.c | 48 ++++++++++++++++++++++++++++++++++--------- 2 files changed, 42 insertions(+), 13 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH V2] ASoC: fsl_asrc: add constraint for the asrc of older version
by S.j. Wang 02 Mar '19

02 Mar '19

There is constraint for the channel number setting on the asrc of older version (e.g. imx35), the channel number should be even, odd number isn't valid. So add this constraint when the asrc of older version is used. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - switch to add constraint in startup function sound/soc/fsl/fsl_asrc.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 528e8b108422..31494f225037 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -445,6 +445,19 @@ struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir) } EXPORT_SYMBOL_GPL(fsl_asrc_get_dma_channel); +static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + + /* channel_bits = 3 means older version on imx35*/ + if (asrc_priv->channel_bits == 3) + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -539,6 +552,7 @@ static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops fsl_asrc_dai_ops = { + .startup = fsl_asrc_dai_startup, .hw_params = fsl_asrc_dai_hw_params, .hw_free = fsl_asrc_dai_hw_free, .trigger = fsl_asrc_dai_trigger, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: cs42xx8: Remove S32_LE in format list
by S.j. Wang 02 Mar '19

02 Mar '19

cs42xx8 is a 24-bit A/D and 24-bit D/A device, so the S32_LE should not be in the supported format list. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..271972337d23 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -32,8 +32,7 @@ #define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_LE) /* codec private data */ struct cs42xx8_priv { -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: core/dapm/topology: minor fixes
by Pierre-Louis Bossart 02 Mar '19

02 Mar '19

3 small patches that have been used for a while but not contributed upstream. The first patch fixes a name mismatch when using the same machine driver on ApolloLake and GeminiLake Chromebooks. The second one fixes a minor mistake in the location of a topology pointer update, and aligns the code on the same pattern. The last one corrects yet another miss with the introduction of the new buffer/src/scheduler soc_dapm elements which were always showing as 'On' Liam Girdwood (1): ASoC: topology: Align tplg pointer increment across all kcontrols Pierre-Louis Bossart (1): ASoC: core: support driver alias names for FE topology overrides Ranjani Sridharan (1): ASoC: dapm: set power_check callback for widgets that shouldnt be always on sound/soc/soc-core.c | 7 +++++-- sound/soc/soc-dapm.c | 7 +++++++ sound/soc/soc-topology.c | 17 +++++++++-------- 3 files changed, 21 insertions(+), 10 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH 3/3] arm64: dts: hi3660-hikey960: Add i2s & sound device
by Pengcheng Li 01 Mar '19

01 Mar '19

From: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Tanglei Han <hantanglei(a)huawei.com> Signed-off-by: Guangke Ji <jiguangke(a)huawei.com> Signed-off-by: Feng Chen <puck.chen(a)hisilicon.com> Signed-off-by: Kaihua Zhong <zhongkaihua(a)huawei.com> Signed-off-by: Jun Chen <chenjun14(a)huawei.com> Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Yiping Xu <xuyiping(a)hisilicon.com> Signed-off-by: Pengcheng Li <lipengcheng8(a)huawei.com> Cc: Wei Xu <xuwei5(a)hisilicon.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> --- arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts | 29 +++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts b/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts index 186251f..457cebc 100644 --- a/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts +++ b/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts @@ -27,6 +27,35 @@ /* rewrite this at bootloader */ reg = <0x0 0x0 0x0 0x0>; }; + + i2s2: hi3660_i2s { + compatible = "hisilicon,hi3660-i2s-1.0"; + reg = <0x0 0xe804f800 0x0 0x400>, + <0x0 0xe804e000 0x0 0x400>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pmx_func &i2s2_cfg_func>; + dmas = <&asp_dmac 18 &asp_dmac 19>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "hikey-hdmi"; + simple-audio-card,format = "i2s"; + + simple-audio-card,bitclock-master = <&sound_master>; + simple-audio-card,frame-master = <&sound_master>; + + sound_master: simple-audio-card,cpu { + sound-dai = <&i2s2>; + }; + + simple-audio-card,codec { + sound-dai = <&adv7533>; + }; + }; + }; &uart5 { -- 2.8.0

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[alsa-devel] [PATCH] ASoC: fsl_asrc: add protection for the asrc of older version
by S.j. Wang 01 Mar '19

01 Mar '19

There is a constraint for the channel number setting on the asrc of older version (e.g. imx35), the channel number should be even, odd number isn't valid. So add protection when the asrc of older version is used. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 528e8b108422..b3b3c5e15ef1 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -109,7 +109,8 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) if (index == ASRC_INVALID_PAIR) { dev_err(dev, "all pairs are busy now\n"); ret = -EBUSY; - } else if (asrc_priv->channel_avail < channels) { + } else if (asrc_priv->channel_avail < channels || + (asrc_priv->channel_bits < 4 && channels % 2 != 0)) { dev_err(dev, "can't afford required channels: %d\n", channels); ret = -EINVAL; } else { -- 1.9.1

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Re: [alsa-devel] [PATCH] ASoC: cs42xx8: Remove S32_LE in format list
by S.j. Wang 01 Mar '19

01 Mar '19

Hi > > cs42xx8 is a 24-bit A/D and 24-bit D/A device, so the S32_LE should > > not be in the supported format list. > > > > Signed-off-by: Shengjiu Wang<shengjiu.wang(a)nxp.com> > > Is the device capable of accepting 32-bit samples, even if it downgrades it to > 24-bit internally? If so, then maybe SNDRV_PCM_FMTBIT_S32_LE should > stay. Yes, the S32_LE can be accepted by device, it seems like that the device only Use the upper 24bit, so the S32_LE result is same as S24_LE, do you think the S32_LE should be remained? Best regards Wang shengjiu

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